/*
 *  DSP.js - a comprehensive digital signal processing  library for javascript
 *
 *  Created by Corban Brook <corbanbrook@gmail.com> on 2010-01-01.
 *  Copyright 2010 Corban Brook. All rights reserved.
 *
 */

////////////////////////////////////////////////////////////////////////////////
//                                  CONSTANTS                                 //
////////////////////////////////////////////////////////////////////////////////

/**
 * DSP is an object which contains general purpose utility functions and constants
 */
var DSP = {
    // Channels
    LEFT:           0,
    RIGHT:          1,
    MIX:            2,

    // Waveforms
    SINE:           1,
    TRIANGLE:       2,
    SAW:            3,
    SQUARE:         4,

    // Filters
    LOWPASS:        0,
    HIGHPASS:       1,
    BANDPASS:       2,
    NOTCH:          3,

    // Window functions
    BARTLETT:       1,
    BARTLETTHANN:   2,
    BLACKMAN:       3,
    COSINE:         4,
    GAUSS:          5,
    HAMMING:        6,
    HANN:           7,
    LANCZOS:        8,
    RECTANGULAR:    9,
    TRIANGULAR:     10,

    // Loop modes
    OFF:            0,
    FW:             1,
    BW:             2,
    FWBW:           3,

    // Math
    TWO_PI:         2*Math.PI
};

// Setup arrays for platforms which do not support byte arrays
function setupTypedArray(name, fallback) {
    // check if TypedArray exists
    // typeof on Minefield and Chrome return function, typeof on Webkit returns object.
    if (typeof this[name] !== "function" && typeof this[name] !== "object") {
        // nope.. check if WebGLArray exists
        if (typeof this[fallback] === "function" && typeof this[fallback] !== "object") {
            this[name] = this[fallback];
        } else {
            // nope.. set as Native JS array
            this[name] = function(obj) {
                if (obj instanceof Array) {
                    return obj;
                } else if (typeof obj === "number") {
                    return new Array(obj);
                }
            };
        }
    }
}

setupTypedArray("Float32Array", "WebGLFloatArray");
setupTypedArray("Int32Array",   "WebGLIntArray");
setupTypedArray("Uint16Array",  "WebGLUnsignedShortArray");
setupTypedArray("Uint8Array",   "WebGLUnsignedByteArray");


////////////////////////////////////////////////////////////////////////////////
//                            DSP UTILITY FUNCTIONS                           //
////////////////////////////////////////////////////////////////////////////////

/**
 * Inverts the phase of a signal
 *
 * @param {Array} buffer A sample buffer
 *
 * @returns The inverted sample buffer
 */
DSP.invert = function(buffer) {
    for (var i = 0, len = buffer.length; i < len; i++) {
        buffer[i] *= -1;
    }

    return buffer;
};

/**
 * Converts split-stereo (dual mono) sample buffers into a stereo interleaved sample buffer
 *
 * @param {Array} left  A sample buffer
 * @param {Array} right A sample buffer
 *
 * @returns The stereo interleaved buffer
 */
DSP.interleave = function(left, right) {
    if (left.length !== right.length) {
        throw "Can not interleave. Channel lengths differ.";
    }

    var stereoInterleaved = new Float32Array(left.length * 2);

    for (var i = 0, len = left.length; i < len; i++) {
        stereoInterleaved[2*i]   = left[i];
        stereoInterleaved[2*i+1] = right[i];
    }

    return stereoInterleaved;
};

/**
 * Converts a stereo-interleaved sample buffer into split-stereo (dual mono) sample buffers
 *
 * @param {Array} buffer A stereo-interleaved sample buffer
 *
 * @returns an Array containing left and right channels
 */
DSP.deinterleave = (function() {
    var left, right, mix, deinterleaveChannel = [];

    deinterleaveChannel[DSP.MIX] = function(buffer) {
        for (var i = 0, len = buffer.length/2; i < len; i++) {
            mix[i] = (buffer[2*i] + buffer[2*i+1]) / 2;
        }
        return mix;
    };

    deinterleaveChannel[DSP.LEFT] = function(buffer) {
        for (var i = 0, len = buffer.length/2; i < len; i++) {
            left[i]  = buffer[2*i];
        }
        return left;
    };

    deinterleaveChannel[DSP.RIGHT] = function(buffer) {
        for (var i = 0, len = buffer.length/2; i < len; i++) {
            right[i]  = buffer[2*i+1];
        }
        return right;
    };

    return function(channel, buffer) {
        left  = left  || new Float32Array(buffer.length/2);
        right = right || new Float32Array(buffer.length/2);
        mix   = mix   || new Float32Array(buffer.length/2);

        if (buffer.length/2 !== left.length) {
            left  = new Float32Array(buffer.length/2);
            right = new Float32Array(buffer.length/2);
            mix   = new Float32Array(buffer.length/2);
        }

        return deinterleaveChannel[channel](buffer);
    };
}());

/**
 * Separates a channel from a stereo-interleaved sample buffer
 *
 * @param {Array}  buffer A stereo-interleaved sample buffer
 * @param {Number} channel A channel constant (LEFT, RIGHT, MIX)
 *
 * @returns an Array containing a signal mono sample buffer
 */
DSP.getChannel = DSP.deinterleave;

/**
 * Helper method (for Reverb) to mix two (interleaved) samplebuffers. It's possible
 * to negate the second buffer while mixing and to perform a volume correction
 * on the final signal.
 *
 * @param {Array} sampleBuffer1 Array containing Float values or a Float32Array
 * @param {Array} sampleBuffer2 Array containing Float values or a Float32Array
 * @param {Boolean} negate When true inverts/flips the audio signal
 * @param {Number} volumeCorrection When you add multiple sample buffers, use this to tame your signal ;)
 *
 * @returns A new Float32Array interleaved buffer.
 */
DSP.mixSampleBuffers = function(sampleBuffer1, sampleBuffer2, negate, volumeCorrection){
    var outputSamples = new Float32Array(sampleBuffer1);

    for(var i = 0; i<sampleBuffer1.length; i++){
        outputSamples[i] += (negate ? -sampleBuffer2[i] : sampleBuffer2[i]) / volumeCorrection;
    }

    return outputSamples;
};

// Biquad filter types
DSP.LPF = 0;                // H(s) = 1 / (s^2 + s/Q + 1)
DSP.HPF = 1;                // H(s) = s^2 / (s^2 + s/Q + 1)
DSP.BPF_CONSTANT_SKIRT = 2; // H(s) = s / (s^2 + s/Q + 1)  (constant skirt gain, peak gain = Q)
DSP.BPF_CONSTANT_PEAK = 3;  // H(s) = (s/Q) / (s^2 + s/Q + 1)      (constant 0 dB peak gain)
DSP.NOTCH = 4;              // H(s) = (s^2 + 1) / (s^2 + s/Q + 1)
DSP.APF = 5;                // H(s) = (s^2 - s/Q + 1) / (s^2 + s/Q + 1)
DSP.PEAKING_EQ = 6;         // H(s) = (s^2 + s*(A/Q) + 1) / (s^2 + s/(A*Q) + 1)
DSP.LOW_SHELF = 7;          // H(s) = A * (s^2 + (sqrt(A)/Q)*s + A)/(A*s^2 + (sqrt(A)/Q)*s + 1)
DSP.HIGH_SHELF = 8;         // H(s) = A * (A*s^2 + (sqrt(A)/Q)*s + 1)/(s^2 + (sqrt(A)/Q)*s + A)

// Biquad filter parameter types
DSP.Q = 1;
DSP.BW = 2; // SHARED with BACKWARDS LOOP MODE
DSP.S = 3;

// Find RMS of signal
DSP.RMS = function(buffer) {
    var total = 0;

    for (var i = 0, n = buffer.length; i < n; i++) {
        total += buffer[i] * buffer[i];
    }

    return Math.sqrt(total / n);
};

// Find Peak of signal
DSP.Peak = function(buffer) {
    var peak = 0;

    for (var i = 0, n = buffer.length; i < n; i++) {
        peak = (Math.abs(buffer[i]) > peak) ? Math.abs(buffer[i]) : peak;
    }

    return peak;
};

// Fourier Transform Module used by DFT, FFT, RFFT
function FourierTransform(bufferSize, sampleRate) {
    this.bufferSize = bufferSize;
    this.sampleRate = sampleRate;
    this.bandwidth  = 2 / bufferSize * sampleRate / 2;

    this.spectrum   = new Float32Array(bufferSize/2);
    this.real       = new Float32Array(bufferSize);
    this.imag       = new Float32Array(bufferSize);

    this.peakBand   = 0;
    this.peak       = 0;

    /**
     * Calculates the *middle* frequency of an FFT band.
     *
     * @param {Number} index The index of the FFT band.
     *
     * @returns The middle frequency in Hz.
     */
    this.getBandFrequency = function(index) {
        return this.bandwidth * index + this.bandwidth / 2;
    };

    this.calculateSpectrum = function() {
        var spectrum  = this.spectrum,
            real      = this.real,
            imag      = this.imag,
            bSi       = 2 / this.bufferSize,
            sqrt      = Math.sqrt,
            rval,
            ival,
            mag;

        for (var i = 0, N = bufferSize/2; i < N; i++) {
            rval = real[i];
            ival = imag[i];
            mag = bSi * sqrt(rval * rval + ival * ival);

            if (mag > this.peak) {
                this.peakBand = i;
                this.peak = mag;
            }

            spectrum[i] = mag;
        }
    };
}

/**
 * DFT is a class for calculating the Discrete Fourier Transform of a signal.
 *
 * @param {Number} bufferSize The size of the sample buffer to be computed
 * @param {Number} sampleRate The sampleRate of the buffer (eg. 44100)
 *
 * @constructor
 */
function DFT(bufferSize, sampleRate) {
    FourierTransform.call(this, bufferSize, sampleRate);

    var N = bufferSize/2 * bufferSize;
    var TWO_PI = 2 * Math.PI;

    this.sinTable = new Float32Array(N);
    this.cosTable = new Float32Array(N);

    for (var i = 0; i < N; i++) {
        this.sinTable[i] = Math.sin(i * TWO_PI / bufferSize);
        this.cosTable[i] = Math.cos(i * TWO_PI / bufferSize);
    }
}

/**
 * Performs a forward transform on the sample buffer.
 * Converts a time domain signal to frequency domain spectra.
 *
 * @param {Array} buffer The sample buffer
 *
 * @returns The frequency spectrum array
 */
DFT.prototype.forward = function(buffer) {
    var real = this.real,
        imag = this.imag,
        rval,
        ival;

    for (var k = 0; k < this.bufferSize/2; k++) {
        rval = 0.0;
        ival = 0.0;

        for (var n = 0; n < buffer.length; n++) {
            rval += this.cosTable[k*n] * buffer[n];
            ival += this.sinTable[k*n] * buffer[n];
        }

        real[k] = rval;
        imag[k] = ival;
    }

    return this.calculateSpectrum();
};


/**
 * FFT is a class for calculating the Discrete Fourier Transform of a signal
 * with the Fast Fourier Transform algorithm.
 *
 * @param {Number} bufferSize The size of the sample buffer to be computed. Must be power of 2
 * @param {Number} sampleRate The sampleRate of the buffer (eg. 44100)
 *
 * @constructor
 */
function FFT(bufferSize, sampleRate) {
    FourierTransform.call(this, bufferSize, sampleRate);

    this.reverseTable = new Uint32Array(bufferSize);

    var limit = 1;
    var bit = bufferSize >> 1;

    var i;

    while (limit < bufferSize) {
        for (i = 0; i < limit; i++) {
            this.reverseTable[i + limit] = this.reverseTable[i] + bit;
        }

        limit = limit << 1;
        bit = bit >> 1;
    }

    this.sinTable = new Float32Array(bufferSize);
    this.cosTable = new Float32Array(bufferSize);

    for (i = 0; i < bufferSize; i++) {
        this.sinTable[i] = Math.sin(-Math.PI/i);
        this.cosTable[i] = Math.cos(-Math.PI/i);
    }
}

/**
 * Performs a forward transform on the sample buffer.
 * Converts a time domain signal to frequency domain spectra.
 *
 * @param {Array} buffer The sample buffer. Buffer Length must be power of 2
 *
 * @returns The frequency spectrum array
 */
FFT.prototype.forward = function(buffer) {
    // Locally scope variables for speed up
    var bufferSize      = this.bufferSize,
        cosTable        = this.cosTable,
        sinTable        = this.sinTable,
        reverseTable    = this.reverseTable,
        real            = this.real,
        imag            = this.imag,
        spectrum        = this.spectrum;

    var k = Math.floor(Math.log(bufferSize) / Math.LN2);

    if (Math.pow(2, k) !== bufferSize) { throw "Invalid buffer size, must be a power of 2."; }
    if (bufferSize !== buffer.length)  { throw "Supplied buffer is not the same size as defined FFT. FFT Size: " + bufferSize + " Buffer Size: " + buffer.length; }

    var halfSize = 1,
        phaseShiftStepReal,
        phaseShiftStepImag,
        currentPhaseShiftReal,
        currentPhaseShiftImag,
        off,
        tr,
        ti,
        tmpReal,
        i;

    for (i = 0; i < bufferSize; i++) {
        real[i] = buffer[reverseTable[i]];
        imag[i] = 0;
    }

    while (halfSize < bufferSize) {
        //phaseShiftStepReal = Math.cos(-Math.PI/halfSize);
        //phaseShiftStepImag = Math.sin(-Math.PI/halfSize);
        phaseShiftStepReal = cosTable[halfSize];
        phaseShiftStepImag = sinTable[halfSize];

        currentPhaseShiftReal = 1;
        currentPhaseShiftImag = 0;

        for (var fftStep = 0; fftStep < halfSize; fftStep++) {
            i = fftStep;

            while (i < bufferSize) {
                off = i + halfSize;
                tr = (currentPhaseShiftReal * real[off]) - (currentPhaseShiftImag * imag[off]);
                ti = (currentPhaseShiftReal * imag[off]) + (currentPhaseShiftImag * real[off]);

                real[off] = real[i] - tr;
                imag[off] = imag[i] - ti;
                real[i] += tr;
                imag[i] += ti;

                i += halfSize << 1;
            }

            tmpReal = currentPhaseShiftReal;
            currentPhaseShiftReal = (tmpReal * phaseShiftStepReal) - (currentPhaseShiftImag * phaseShiftStepImag);
            currentPhaseShiftImag = (tmpReal * phaseShiftStepImag) + (currentPhaseShiftImag * phaseShiftStepReal);
        }

        halfSize = halfSize << 1;
    }

    return this.calculateSpectrum();
};

FFT.prototype.inverse = function(real, imag) {
    // Locally scope variables for speed up
    var bufferSize      = this.bufferSize,
        cosTable        = this.cosTable,
        sinTable        = this.sinTable,
        reverseTable    = this.reverseTable,
        spectrum        = this.spectrum;

    real = real || this.real;
    imag = imag || this.imag;

    var halfSize = 1,
        phaseShiftStepReal,
        phaseShiftStepImag,
        currentPhaseShiftReal,
        currentPhaseShiftImag,
        off,
        tr,
        ti,
        tmpReal,
        i;

    for (i = 0; i < bufferSize; i++) {
        imag[i] *= -1;
    }

    var revReal = new Float32Array(bufferSize);
    var revImag = new Float32Array(bufferSize);

    for (i = 0; i < real.length; i++) {
        revReal[i] = real[reverseTable[i]];
        revImag[i] = imag[reverseTable[i]];
    }

    real = revReal;
    imag = revImag;

    while (halfSize < bufferSize) {
        phaseShiftStepReal = cosTable[halfSize];
        phaseShiftStepImag = sinTable[halfSize];
        currentPhaseShiftReal = 1;
        currentPhaseShiftImag = 0;

        for (var fftStep = 0; fftStep < halfSize; fftStep++) {
            i = fftStep;

            while (i < bufferSize) {
                off = i + halfSize;
                tr = (currentPhaseShiftReal * real[off]) - (currentPhaseShiftImag * imag[off]);
                ti = (currentPhaseShiftReal * imag[off]) + (currentPhaseShiftImag * real[off]);

                real[off] = real[i] - tr;
                imag[off] = imag[i] - ti;
                real[i] += tr;
                imag[i] += ti;

                i += halfSize << 1;
            }

            tmpReal = currentPhaseShiftReal;
            currentPhaseShiftReal = (tmpReal * phaseShiftStepReal) - (currentPhaseShiftImag * phaseShiftStepImag);
            currentPhaseShiftImag = (tmpReal * phaseShiftStepImag) + (currentPhaseShiftImag * phaseShiftStepReal);
        }

        halfSize = halfSize << 1;
    }

    var buffer = new Float32Array(bufferSize); // this should be reused instead
    for (i = 0; i < bufferSize; i++) {
        buffer[i] = real[i] / bufferSize;
    }

    return buffer;
};

/**
 * RFFT is a class for calculating the Discrete Fourier Transform of a signal
 * with the Fast Fourier Transform algorithm.
 *
 * This method currently only contains a forward transform but is highly optimized.
 *
 * @param {Number} bufferSize The size of the sample buffer to be computed. Must be power of 2
 * @param {Number} sampleRate The sampleRate of the buffer (eg. 44100)
 *
 * @constructor
 */

// lookup tables don't really gain us any speed, but they do increase
// cache footprint, so don't use them in here

// also we don't use sepearate arrays for real/imaginary parts

// this one a little more than twice as fast as the one in FFT
// however I only did the forward transform

// the rest of this was translated from C, see http://www.jjj.de/fxt/
// this is the real split radix FFT

function RFFT(bufferSize, sampleRate) {
    FourierTransform.call(this, bufferSize, sampleRate);

    this.trans = new Float32Array(bufferSize);

    this.reverseTable = new Uint32Array(bufferSize);

    // don't use a lookup table to do the permute, use this instead
    this.reverseBinPermute = function (dest, source) {
        var bufferSize  = this.bufferSize,
            halfSize    = bufferSize >>> 1,
            nm1         = bufferSize - 1,
            i = 1, r = 0, h;

        dest[0] = source[0];

        do {
            r += halfSize;
            dest[i] = source[r];
            dest[r] = source[i];

            i++;

            h = halfSize << 1;
            while (h = h >> 1, !((r ^= h) & h));

            if (r >= i) {
                dest[i]     = source[r];
                dest[r]     = source[i];

                dest[nm1-i] = source[nm1-r];
                dest[nm1-r] = source[nm1-i];
            }
            i++;
        } while (i < halfSize);
        dest[nm1] = source[nm1];
    };

    this.generateReverseTable = function () {
        var bufferSize  = this.bufferSize,
            halfSize    = bufferSize >>> 1,
            nm1         = bufferSize - 1,
            i = 1, r = 0, h;

        this.reverseTable[0] = 0;

        do {
            r += halfSize;

            this.reverseTable[i] = r;
            this.reverseTable[r] = i;

            i++;

            h = halfSize << 1;
            while (h = h >> 1, !((r ^= h) & h));

            if (r >= i) {
                this.reverseTable[i] = r;
                this.reverseTable[r] = i;

                this.reverseTable[nm1-i] = nm1-r;
                this.reverseTable[nm1-r] = nm1-i;
            }
            i++;
        } while (i < halfSize);

        this.reverseTable[nm1] = nm1;
    };

    this.generateReverseTable();
}


// Ordering of output:
//
// trans[0]     = re[0] (==zero frequency, purely real)
// trans[1]     = re[1]
//             ...
// trans[n/2-1] = re[n/2-1]
// trans[n/2]   = re[n/2]    (==nyquist frequency, purely real)
//
// trans[n/2+1] = im[n/2-1]
// trans[n/2+2] = im[n/2-2]
//             ...
// trans[n-1]   = im[1]

RFFT.prototype.forward = function(buffer) {
    var n         = this.bufferSize,
        spectrum  = this.spectrum,
        x         = this.trans,
        TWO_PI    = 2*Math.PI,
        sqrt      = Math.sqrt,
        i         = n >>> 1,
        bSi       = 2 / n,
        n2, n4, n8, nn,
        t1, t2, t3, t4,
        i1, i2, i3, i4, i5, i6, i7, i8,
        st1, cc1, ss1, cc3, ss3,
        e,
        a,
        rval, ival, mag;

    this.reverseBinPermute(x, buffer);

    /*
     var reverseTable = this.reverseTable;

     for (var k = 0, len = reverseTable.length; k < len; k++) {
     x[k] = buffer[reverseTable[k]];
     }
     */

    for (var ix = 0, id = 4; ix < n; id *= 4) {
        for (var i0 = ix; i0 < n; i0 += id) {
            //sumdiff(x[i0], x[i0+1]); // {a, b}  <--| {a+b, a-b}
            st1 = x[i0] - x[i0+1];
            x[i0] += x[i0+1];
            x[i0+1] = st1;
        }
        ix = 2*(id-1);
    }

    n2 = 2;
    nn = n >>> 1;

    while((nn = nn >>> 1)) {
        ix = 0;
        n2 = n2 << 1;
        id = n2 << 1;
        n4 = n2 >>> 2;
        n8 = n2 >>> 3;
        do {
            if(n4 !== 1) {
                for(i0 = ix; i0 < n; i0 += id) {
                    i1 = i0;
                    i2 = i1 + n4;
                    i3 = i2 + n4;
                    i4 = i3 + n4;

                    //diffsum3_r(x[i3], x[i4], t1); // {a, b, s} <--| {a, b-a, a+b}
                    t1 = x[i3] + x[i4];
                    x[i4] -= x[i3];
                    //sumdiff3(x[i1], t1, x[i3]);   // {a, b, d} <--| {a+b, b, a-b}
                    x[i3] = x[i1] - t1;
                    x[i1] += t1;

                    i1 += n8;
                    i2 += n8;
                    i3 += n8;
                    i4 += n8;

                    //sumdiff(x[i3], x[i4], t1, t2); // {s, d}  <--| {a+b, a-b}
                    t1 = x[i3] + x[i4];
                    t2 = x[i3] - x[i4];

                    t1 = -t1 * Math.SQRT1_2;
                    t2 *= Math.SQRT1_2;

                    // sumdiff(t1, x[i2], x[i4], x[i3]); // {s, d}  <--| {a+b, a-b}
                    st1 = x[i2];
                    x[i4] = t1 + st1;
                    x[i3] = t1 - st1;

                    //sumdiff3(x[i1], t2, x[i2]); // {a, b, d} <--| {a+b, b, a-b}
                    x[i2] = x[i1] - t2;
                    x[i1] += t2;
                }
            } else {
                for(i0 = ix; i0 < n; i0 += id) {
                    i1 = i0;
                    i2 = i1 + n4;
                    i3 = i2 + n4;
                    i4 = i3 + n4;

                    //diffsum3_r(x[i3], x[i4], t1); // {a, b, s} <--| {a, b-a, a+b}
                    t1 = x[i3] + x[i4];
                    x[i4] -= x[i3];

                    //sumdiff3(x[i1], t1, x[i3]);   // {a, b, d} <--| {a+b, b, a-b}
                    x[i3] = x[i1] - t1;
                    x[i1] += t1;
                }
            }

            ix = (id << 1) - n2;
            id = id << 2;
        } while (ix < n);

        e = TWO_PI / n2;

        for (var j = 1; j < n8; j++) {
            a = j * e;
            ss1 = Math.sin(a);
            cc1 = Math.cos(a);

            //ss3 = sin(3*a); cc3 = cos(3*a);
            cc3 = 4*cc1*(cc1*cc1-0.75);
            ss3 = 4*ss1*(0.75-ss1*ss1);

            ix = 0; id = n2 << 1;
            do {
                for (i0 = ix; i0 < n; i0 += id) {
                    i1 = i0 + j;
                    i2 = i1 + n4;
                    i3 = i2 + n4;
                    i4 = i3 + n4;

                    i5 = i0 + n4 - j;
                    i6 = i5 + n4;
                    i7 = i6 + n4;
                    i8 = i7 + n4;

                    //cmult(c, s, x, y, &u, &v)
                    //cmult(cc1, ss1, x[i7], x[i3], t2, t1); // {u,v} <--| {x*c-y*s, x*s+y*c}
                    t2 = x[i7]*cc1 - x[i3]*ss1;
                    t1 = x[i7]*ss1 + x[i3]*cc1;

                    //cmult(cc3, ss3, x[i8], x[i4], t4, t3);
                    t4 = x[i8]*cc3 - x[i4]*ss3;
                    t3 = x[i8]*ss3 + x[i4]*cc3;

                    //sumdiff(t2, t4);   // {a, b} <--| {a+b, a-b}
                    st1 = t2 - t4;
                    t2 += t4;
                    t4 = st1;

                    //sumdiff(t2, x[i6], x[i8], x[i3]); // {s, d}  <--| {a+b, a-b}
                    //st1 = x[i6]; x[i8] = t2 + st1; x[i3] = t2 - st1;
                    x[i8] = t2 + x[i6];
                    x[i3] = t2 - x[i6];

                    //sumdiff_r(t1, t3); // {a, b} <--| {a+b, b-a}
                    st1 = t3 - t1;
                    t1 += t3;
                    t3 = st1;

                    //sumdiff(t3, x[i2], x[i4], x[i7]); // {s, d}  <--| {a+b, a-b}
                    //st1 = x[i2]; x[i4] = t3 + st1; x[i7] = t3 - st1;
                    x[i4] = t3 + x[i2];
                    x[i7] = t3 - x[i2];

                    //sumdiff3(x[i1], t1, x[i6]);   // {a, b, d} <--| {a+b, b, a-b}
                    x[i6] = x[i1] - t1;
                    x[i1] += t1;

                    //diffsum3_r(t4, x[i5], x[i2]); // {a, b, s} <--| {a, b-a, a+b}
                    x[i2] = t4 + x[i5];
                    x[i5] -= t4;
                }

                ix = (id << 1) - n2;
                id = id << 2;

            } while (ix < n);
        }
    }

    while (--i) {
        rval = x[i];
        ival = x[n-i-1];
        mag = bSi * sqrt(rval * rval + ival * ival);

        if (mag > this.peak) {
            this.peakBand = i;
            this.peak = mag;
        }

        spectrum[i] = mag;
    }

    spectrum[0] = bSi * x[0];

    return spectrum;
};

function Sampler(file, bufferSize, sampleRate, playStart, playEnd, loopStart, loopEnd, loopMode) {
    this.file = file;
    this.bufferSize = bufferSize;
    this.sampleRate = sampleRate;
    this.playStart  = playStart || 0; // 0%
    this.playEnd    = playEnd   || 1; // 100%
    this.loopStart  = loopStart || 0;
    this.loopEnd    = loopEnd   || 1;
    this.loopMode   = loopMode  || DSP.OFF;
    this.loaded     = false;
    this.samples    = [];
    this.signal     = new Float32Array(bufferSize);
    this.frameCount = 0;
    this.envelope   = null;
    this.amplitude  = 1;
    this.rootFrequency = 110; // A2 110
    this.frequency  = 550;
    this.step       = this.frequency / this.rootFrequency;
    this.duration   = 0;
    this.samplesProcessed = 0;
    this.playhead   = 0;

    var audio = /* new Audio();*/ document.createElement("AUDIO");
    var self = this;

    this.loadSamples = function(event) {
        var buffer = DSP.getChannel(DSP.MIX, event.frameBuffer);
        for ( var i = 0; i < buffer.length; i++) {
            self.samples.push(buffer[i]);
        }
    };

    this.loadComplete = function() {
        // convert flexible js array into a fast typed array
        self.samples = new Float32Array(self.samples);
        self.loaded = true;
    };

    this.loadMetaData = function() {
        self.duration = audio.duration;
    };

    audio.addEventListener("MozAudioAvailable", this.loadSamples, false);
    audio.addEventListener("loadedmetadata", this.loadMetaData, false);
    audio.addEventListener("ended", this.loadComplete, false);
    audio.muted = true;
    audio.src = file;
    audio.play();
}

Sampler.prototype.applyEnvelope = function() {
    this.envelope.process(this.signal);
    return this.signal;
};

Sampler.prototype.generate = function() {
    var frameOffset = this.frameCount * this.bufferSize;

    var loopWidth = this.playEnd * this.samples.length - this.playStart * this.samples.length;
    var playStartSamples = this.playStart * this.samples.length; // ie 0.5 -> 50% of the length
    var playEndSamples = this.playEnd * this.samples.length; // ie 0.5 -> 50% of the length
    var offset;

    for ( var i = 0; i < this.bufferSize; i++ ) {
        switch (this.loopMode) {
            case DSP.OFF:
                this.playhead = Math.round(this.samplesProcessed * this.step + playStartSamples);
                if (this.playhead < (this.playEnd * this.samples.length) ) {
                    this.signal[i] = this.samples[this.playhead] * this.amplitude;
                } else {
                    this.signal[i] = 0;
                }
                break;

            case DSP.FW:
                this.playhead = Math.round((this.samplesProcessed * this.step) % loopWidth + playStartSamples);
                if (this.playhead < (this.playEnd * this.samples.length) ) {
                    this.signal[i] = this.samples[this.playhead] * this.amplitude;
                }
                break;

            case DSP.BW:
                this.playhead = playEndSamples - Math.round((this.samplesProcessed * this.step) % loopWidth);
                if (this.playhead < (this.playEnd * this.samples.length) ) {
                    this.signal[i] = this.samples[this.playhead] * this.amplitude;
                }
                break;

            case DSP.FWBW:
                if ( Math.floor(this.samplesProcessed * this.step / loopWidth) % 2 === 0 ) {
                    this.playhead = Math.round((this.samplesProcessed * this.step) % loopWidth + playStartSamples);
                } else {
                    this.playhead = playEndSamples - Math.round((this.samplesProcessed * this.step) % loopWidth);
                }
                if (this.playhead < (this.playEnd * this.samples.length) ) {
                    this.signal[i] = this.samples[this.playhead] * this.amplitude;
                }
                break;
        }
        this.samplesProcessed++;
    }

    this.frameCount++;

    return this.signal;
};

Sampler.prototype.setFreq = function(frequency) {
    var totalProcessed = this.samplesProcessed * this.step;
    this.frequency = frequency;
    this.step = this.frequency / this.rootFrequency;
    this.samplesProcessed = Math.round(totalProcessed/this.step);
};

Sampler.prototype.reset = function() {
    this.samplesProcessed = 0;
    this.playhead = 0;
};

/**
 * Oscillator class for generating and modifying signals
 *
 * @param {Number} type       A waveform constant (eg. DSP.SINE)
 * @param {Number} frequency  Initial frequency of the signal
 * @param {Number} amplitude  Initial amplitude of the signal
 * @param {Number} bufferSize Size of the sample buffer to generate
 * @param {Number} sampleRate The sample rate of the signal
 *
 * @contructor
 */
function Oscillator(type, frequency, amplitude, bufferSize, sampleRate) {
    this.frequency  = frequency;
    this.amplitude  = amplitude;
    this.bufferSize = bufferSize;
    this.sampleRate = sampleRate;
    //this.pulseWidth = pulseWidth;
    this.frameCount = 0;

    this.waveTableLength = 2048;

    this.cyclesPerSample = frequency / sampleRate;

    this.signal = new Float32Array(bufferSize);
    this.envelope = null;

    switch(parseInt(type, 10)) {
        case DSP.TRIANGLE:
            this.func = Oscillator.Triangle;
            break;

        case DSP.SAW:
            this.func = Oscillator.Saw;
            break;

        case DSP.SQUARE:
            this.func = Oscillator.Square;
            break;

        default:
        case DSP.SINE:
            this.func = Oscillator.Sine;
            break;
    }

    this.generateWaveTable = function() {
        Oscillator.waveTable[this.func] = new Float32Array(2048);
        var waveTableTime = this.waveTableLength / this.sampleRate;
        var waveTableHz = 1 / waveTableTime;

        for (var i = 0; i < this.waveTableLength; i++) {
            Oscillator.waveTable[this.func][i] = this.func(i * waveTableHz/this.sampleRate);
        }
    };

    if ( typeof Oscillator.waveTable === 'undefined' ) {
        Oscillator.waveTable = {};
    }

    if ( typeof Oscillator.waveTable[this.func] === 'undefined' ) {
        this.generateWaveTable();
    }

    this.waveTable = Oscillator.waveTable[this.func];
}

/**
 * Set the amplitude of the signal
 *
 * @param {Number} amplitude The amplitude of the signal (between 0 and 1)
 */
Oscillator.prototype.setAmp = function(amplitude) {
    if (amplitude >= 0 && amplitude <= 1) {
        this.amplitude = amplitude;
    } else {
        throw "Amplitude out of range (0..1).";
    }
};

/**
 * Set the frequency of the signal
 *
 * @param {Number} frequency The frequency of the signal
 */
Oscillator.prototype.setFreq = function(frequency) {
    this.frequency = frequency;
    this.cyclesPerSample = frequency / this.sampleRate;
};

// Add an oscillator
Oscillator.prototype.add = function(oscillator) {
    for ( var i = 0; i < this.bufferSize; i++ ) {
        //this.signal[i] += oscillator.valueAt(i);
        this.signal[i] += oscillator.signal[i];
    }

    return this.signal;
};

// Add a signal to the current generated osc signal
Oscillator.prototype.addSignal = function(signal) {
    for ( var i = 0; i < signal.length; i++ ) {
        if ( i >= this.bufferSize ) {
            break;
        }
        this.signal[i] += signal[i];

        /*
         // Constrain amplitude
         if ( this.signal[i] > 1 ) {
         this.signal[i] = 1;
         } else if ( this.signal[i] < -1 ) {
         this.signal[i] = -1;
         }
         */
    }
    return this.signal;
};

// Add an envelope to the oscillator
Oscillator.prototype.addEnvelope = function(envelope) {
    this.envelope = envelope;
};

Oscillator.prototype.applyEnvelope = function() {
    this.envelope.process(this.signal);
};

Oscillator.prototype.valueAt = function(offset) {
    return this.waveTable[offset % this.waveTableLength];
};

Oscillator.prototype.generate = function() {
    var frameOffset = this.frameCount * this.bufferSize;
    var step = this.waveTableLength * this.frequency / this.sampleRate;
    var offset;

    for ( var i = 0; i < this.bufferSize; i++ ) {
        //var step = (frameOffset + i) * this.cyclesPerSample % 1;
        //this.signal[i] = this.func(step) * this.amplitude;
        //this.signal[i] = this.valueAt(Math.round((frameOffset + i) * step)) * this.amplitude;
        offset = Math.round((frameOffset + i) * step);
        this.signal[i] = this.waveTable[offset % this.waveTableLength] * this.amplitude;
    }

    this.frameCount++;

    return this.signal;
};

Oscillator.Sine = function(step) {
    return Math.sin(DSP.TWO_PI * step);
};

Oscillator.Square = function(step) {
    return step < 0.5 ? 1 : -1;
};

Oscillator.Saw = function(step) {
    return 2 * (step - Math.round(step));
};

Oscillator.Triangle = function(step) {
    return 1 - 4 * Math.abs(Math.round(step) - step);
};

Oscillator.Pulse = function(step) {
    // stub
};

function ADSR(attackLength, decayLength, sustainLevel, sustainLength, releaseLength, sampleRate) {
    this.sampleRate = sampleRate;
    // Length in seconds
    this.attackLength  = attackLength;
    this.decayLength   = decayLength;
    this.sustainLevel  = sustainLevel;
    this.sustainLength = sustainLength;
    this.releaseLength = releaseLength;
    this.sampleRate    = sampleRate;

    // Length in samples
    this.attackSamples  = attackLength  * sampleRate;
    this.decaySamples   = decayLength   * sampleRate;
    this.sustainSamples = sustainLength * sampleRate;
    this.releaseSamples = releaseLength * sampleRate;

    // Updates the envelope sample positions
    this.update = function() {
        this.attack         =                this.attackSamples;
        this.decay          = this.attack  + this.decaySamples;
        this.sustain        = this.decay   + this.sustainSamples;
        this.release        = this.sustain + this.releaseSamples;
    };

    this.update();

    this.samplesProcessed = 0;
}

ADSR.prototype.noteOn = function() {
    this.samplesProcessed = 0;
    this.sustainSamples = this.sustainLength * this.sampleRate;
    this.update();
};

// Send a note off when using a sustain of infinity to let the envelope enter the release phase
ADSR.prototype.noteOff = function() {
    this.sustainSamples = this.samplesProcessed - this.decaySamples;
    this.update();
};

ADSR.prototype.processSample = function(sample) {
    var amplitude = 0;

    if ( this.samplesProcessed <= this.attack ) {
        amplitude = 0 + (1 - 0) * ((this.samplesProcessed - 0) / (this.attack - 0));
    } else if ( this.samplesProcessed > this.attack && this.samplesProcessed <= this.decay ) {
        amplitude = 1 + (this.sustainLevel - 1) * ((this.samplesProcessed - this.attack) / (this.decay - this.attack));
    } else if ( this.samplesProcessed > this.decay && this.samplesProcessed <= this.sustain ) {
        amplitude = this.sustainLevel;
    } else if ( this.samplesProcessed > this.sustain && this.samplesProcessed <= this.release ) {
        amplitude = this.sustainLevel + (0 - this.sustainLevel) * ((this.samplesProcessed - this.sustain) / (this.release - this.sustain));
    }

    return sample * amplitude;
};

ADSR.prototype.value = function() {
    var amplitude = 0;

    if ( this.samplesProcessed <= this.attack ) {
        amplitude = 0 + (1 - 0) * ((this.samplesProcessed - 0) / (this.attack - 0));
    } else if ( this.samplesProcessed > this.attack && this.samplesProcessed <= this.decay ) {
        amplitude = 1 + (this.sustainLevel - 1) * ((this.samplesProcessed - this.attack) / (this.decay - this.attack));
    } else if ( this.samplesProcessed > this.decay && this.samplesProcessed <= this.sustain ) {
        amplitude = this.sustainLevel;
    } else if ( this.samplesProcessed > this.sustain && this.samplesProcessed <= this.release ) {
        amplitude = this.sustainLevel + (0 - this.sustainLevel) * ((this.samplesProcessed - this.sustain) / (this.release - this.sustain));
    }

    return amplitude;
};

ADSR.prototype.process = function(buffer) {
    for ( var i = 0; i < buffer.length; i++ ) {
        buffer[i] *= this.value();

        this.samplesProcessed++;
    }

    return buffer;
};


ADSR.prototype.isActive = function() {
    if ( this.samplesProcessed > this.release || this.samplesProcessed === -1 ) {
        return false;
    } else {
        return true;
    }
};

ADSR.prototype.disable = function() {
    this.samplesProcessed = -1;
};

function IIRFilter(type, cutoff, resonance, sampleRate) {
    this.sampleRate = sampleRate;

    switch(type) {
        case DSP.LOWPASS:
        case DSP.LP12:
            this.func = new IIRFilter.LP12(cutoff, resonance, sampleRate);
            break;
    }
}

IIRFilter.prototype.__defineGetter__('cutoff',
    function() {
        return this.func.cutoff;
    }
);

IIRFilter.prototype.__defineGetter__('resonance',
    function() {
        return this.func.resonance;
    }
);

IIRFilter.prototype.set = function(cutoff, resonance) {
    this.func.calcCoeff(cutoff, resonance);
};

IIRFilter.prototype.process = function(buffer) {
    this.func.process(buffer);
};

// Add an envelope to the filter
IIRFilter.prototype.addEnvelope = function(envelope) {
    if ( envelope instanceof ADSR ) {
        this.func.addEnvelope(envelope);
    } else {
        throw "Not an envelope.";
    }
};

IIRFilter.LP12 = function(cutoff, resonance, sampleRate) {
    this.sampleRate = sampleRate;
    this.vibraPos   = 0;
    this.vibraSpeed = 0;
    this.envelope = false;

    this.calcCoeff = function(cutoff, resonance) {
        this.w = 2.0 * Math.PI * cutoff / this.sampleRate;
        this.q = 1.0 - this.w / (2.0 * (resonance + 0.5 / (1.0 + this.w)) + this.w - 2.0);
        this.r = this.q * this.q;
        this.c = this.r + 1.0 - 2.0 * Math.cos(this.w) * this.q;

        this.cutoff = cutoff;
        this.resonance = resonance;
    };

    this.calcCoeff(cutoff, resonance);

    this.process = function(buffer) {
        for ( var i = 0; i < buffer.length; i++ ) {
            this.vibraSpeed += (buffer[i] - this.vibraPos) * this.c;
            this.vibraPos   += this.vibraSpeed;
            this.vibraSpeed *= this.r;

            /*
             var temp = this.vibraPos;

             if ( temp > 1.0 ) {
             temp = 1.0;
             } else if ( temp < -1.0 ) {
             temp = -1.0;
             } else if ( temp != temp ) {
             temp = 1;
             }

             buffer[i] = temp;
             */

            if (this.envelope) {
                buffer[i] = (buffer[i] * (1 - this.envelope.value())) + (this.vibraPos * this.envelope.value());
                this.envelope.samplesProcessed++;
            } else {
                buffer[i] = this.vibraPos;
            }
        }
    };
};

IIRFilter.LP12.prototype.addEnvelope = function(envelope) {
    this.envelope = envelope;
};

function IIRFilter2(type, cutoff, resonance, sampleRate) {
    this.type = type;
    this.cutoff = cutoff;
    this.resonance = resonance;
    this.sampleRate = sampleRate;

    this.f = Float32Array(4);
    this.f[0] = 0.0; // lp
    this.f[1] = 0.0; // hp
    this.f[2] = 0.0; // bp
    this.f[3] = 0.0; // br

    this.calcCoeff = function(cutoff, resonance) {
        this.freq = 2 * Math.sin(Math.PI * Math.min(0.25, cutoff/(this.sampleRate*2)));
        this.damp = Math.min(2 * (1 - Math.pow(resonance, 0.25)), Math.min(2, 2/this.freq - this.freq * 0.5));
    };

    this.calcCoeff(cutoff, resonance);
}

IIRFilter2.prototype.process = function(buffer) {
    var input, output;
    var f = this.f;

    for ( var i = 0; i < buffer.length; i++ ) {
        input = buffer[i];

        // first pass
        f[3] = input - this.damp * f[2];
        f[0] = f[0] + this.freq * f[2];
        f[1] = f[3] - f[0];
        f[2] = this.freq * f[1] + f[2];
        output = 0.5 * f[this.type];

        // second pass
        f[3] = input - this.damp * f[2];
        f[0] = f[0] + this.freq * f[2];
        f[1] = f[3] - f[0];
        f[2] = this.freq * f[1] + f[2];
        output += 0.5 * f[this.type];

        if (this.envelope) {
            buffer[i] = (buffer[i] * (1 - this.envelope.value())) + (output * this.envelope.value());
            this.envelope.samplesProcessed++;
        } else {
            buffer[i] = output;
        }
    }
};

IIRFilter2.prototype.addEnvelope = function(envelope) {
    if ( envelope instanceof ADSR ) {
        this.envelope = envelope;
    } else {
        throw "This is not an envelope.";
    }
};

IIRFilter2.prototype.set = function(cutoff, resonance) {
    this.calcCoeff(cutoff, resonance);
};



function WindowFunction(type, alpha) {
    this.alpha = alpha;

    switch(type) {
        case DSP.BARTLETT:
            this.func = WindowFunction.Bartlett;
            break;

        case DSP.BARTLETTHANN:
            this.func = WindowFunction.BartlettHann;
            break;

        case DSP.BLACKMAN:
            this.func = WindowFunction.Blackman;
            this.alpha = this.alpha || 0.16;
            break;

        case DSP.COSINE:
            this.func = WindowFunction.Cosine;
            break;

        case DSP.GAUSS:
            this.func = WindowFunction.Gauss;
            this.alpha = this.alpha || 0.25;
            break;

        case DSP.HAMMING:
            this.func = WindowFunction.Hamming;
            break;

        case DSP.HANN:
            this.func = WindowFunction.Hann;
            break;

        case DSP.LANCZOS:
            this.func = WindowFunction.Lanczoz;
            break;

        case DSP.RECTANGULAR:
            this.func = WindowFunction.Rectangular;
            break;

        case DSP.TRIANGULAR:
            this.func = WindowFunction.Triangular;
            break;
    }
}

WindowFunction.prototype.process = function(buffer) {
    var length = buffer.length;
    for ( var i = 0; i < length; i++ ) {
        buffer[i] *= this.func(length, i, this.alpha);
    }
    return buffer;
};

WindowFunction.Bartlett = function(length, index) {
    return 2 / (length - 1) * ((length - 1) / 2 - Math.abs(index - (length - 1) / 2));
};

WindowFunction.BartlettHann = function(length, index) {
    return 0.62 - 0.48 * Math.abs(index / (length - 1) - 0.5) - 0.38 * Math.cos(DSP.TWO_PI * index / (length - 1));
};

WindowFunction.Blackman = function(length, index, alpha) {
    var a0 = (1 - alpha) / 2;
    var a1 = 0.5;
    var a2 = alpha / 2;

    return a0 - a1 * Math.cos(DSP.TWO_PI * index / (length - 1)) + a2 * Math.cos(4 * Math.PI * index / (length - 1));
};

WindowFunction.Cosine = function(length, index) {
    return Math.cos(Math.PI * index / (length - 1) - Math.PI / 2);
};

WindowFunction.Gauss = function(length, index, alpha) {
    return Math.pow(Math.E, -0.5 * Math.pow((index - (length - 1) / 2) / (alpha * (length - 1) / 2), 2));
};

WindowFunction.Hamming = function(length, index) {
    return 0.54 - 0.46 * Math.cos(DSP.TWO_PI * index / (length - 1));
};

WindowFunction.Hann = function(length, index) {
    return 0.5 * (1 - Math.cos(DSP.TWO_PI * index / (length - 1)));
};

WindowFunction.Lanczos = function(length, index) {
    var x = 2 * index / (length - 1) - 1;
    return Math.sin(Math.PI * x) / (Math.PI * x);
};

WindowFunction.Rectangular = function(length, index) {
    return 1;
};

WindowFunction.Triangular = function(length, index) {
    return 2 / length * (length / 2 - Math.abs(index - (length - 1) / 2));
};

function sinh (arg) {
    // Returns the hyperbolic sine of the number, defined as (exp(number) - exp(-number))/2
    //
    // version: 1004.2314
    // discuss at: http://phpjs.org/functions/sinh    // +   original by: Onno Marsman
    // *     example 1: sinh(-0.9834330348825909);
    // *     returns 1: -1.1497971402636502
    return (Math.exp(arg) - Math.exp(-arg))/2;
}

/*
 *  Biquad filter
 *
 *  Created by Ricard Marxer <email@ricardmarxer.com> on 2010-05-23.
 *  Copyright 2010 Ricard Marxer. All rights reserved.
 *
 */
// Implementation based on:
// http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt
function Biquad(type, sampleRate) {
    this.Fs = sampleRate;
    this.type = type;  // type of the filter
    this.parameterType = DSP.Q; // type of the parameter

    this.x_1_l = 0;
    this.x_2_l = 0;
    this.y_1_l = 0;
    this.y_2_l = 0;

    this.x_1_r = 0;
    this.x_2_r = 0;
    this.y_1_r = 0;
    this.y_2_r = 0;

    this.b0 = 1;
    this.a0 = 1;

    this.b1 = 0;
    this.a1 = 0;

    this.b2 = 0;
    this.a2 = 0;

    this.b0a0 = this.b0 / this.a0;
    this.b1a0 = this.b1 / this.a0;
    this.b2a0 = this.b2 / this.a0;
    this.a1a0 = this.a1 / this.a0;
    this.a2a0 = this.a2 / this.a0;

    this.f0 = 3000;   // "wherever it's happenin', man."  Center Frequency or
    // Corner Frequency, or shelf midpoint frequency, depending
    // on which filter type.  The "significant frequency".

    this.dBgain = 12; // used only for peaking and shelving filters

    this.Q = 1;       // the EE kind of definition, except for peakingEQ in which A*Q is
    // the classic EE Q.  That adjustment in definition was made so that
    // a boost of N dB followed by a cut of N dB for identical Q and
    // f0/Fs results in a precisely flat unity gain filter or "wire".

    this.BW = -3;     // the bandwidth in octaves (between -3 dB frequencies for BPF
    // and notch or between midpoint (dBgain/2) gain frequencies for
    // peaking EQ

    this.S = 1;       // a "shelf slope" parameter (for shelving EQ only).  When S = 1,
    // the shelf slope is as steep as it can be and remain monotonically
    // increasing or decreasing gain with frequency.  The shelf slope, in
    // dB/octave, remains proportional to S for all other values for a
    // fixed f0/Fs and dBgain.

    this.coefficients = function() {
        var b = [this.b0, this.b1, this.b2];
        var a = [this.a0, this.a1, this.a2];
        return {b: b, a:a};
    };

    this.setFilterType = function(type) {
        this.type = type;
        this.recalculateCoefficients();
    };

    this.setSampleRate = function(rate) {
        this.Fs = rate;
        this.recalculateCoefficients();
    };

    this.setQ = function(q) {
        this.parameterType = DSP.Q;
        this.Q = Math.max(Math.min(q, 115.0), 0.001);
        this.recalculateCoefficients();
    };

    this.setBW = function(bw) {
        this.parameterType = DSP.BW;
        this.BW = bw;
        this.recalculateCoefficients();
    };

    this.setS = function(s) {
        this.parameterType = DSP.S;
        this.S = Math.max(Math.min(s, 5.0), 0.0001);
        this.recalculateCoefficients();
    };

    this.setF0 = function(freq) {
        this.f0 = freq;
        this.recalculateCoefficients();
    };

    this.setDbGain = function(g) {
        this.dBgain = g;
        this.recalculateCoefficients();
    };

    this.recalculateCoefficients = function() {
        var A;
        if (type === DSP.PEAKING_EQ || type === DSP.LOW_SHELF || type === DSP.HIGH_SHELF ) {
            A = Math.pow(10, (this.dBgain/40));  // for peaking and shelving EQ filters only
        } else {
            A  = Math.sqrt( Math.pow(10, (this.dBgain/20)) );
        }

        var w0 = DSP.TWO_PI * this.f0 / this.Fs;

        var cosw0 = Math.cos(w0);
        var sinw0 = Math.sin(w0);

        var alpha = 0;

        switch (this.parameterType) {
            case DSP.Q:
                alpha = sinw0/(2*this.Q);
                break;

            case DSP.BW:
                alpha = sinw0 * sinh( Math.LN2/2 * this.BW * w0/sinw0 );
                break;

            case DSP.S:
                alpha = sinw0/2 * Math.sqrt( (A + 1/A)*(1/this.S - 1) + 2 );
                break;
        }

        /**
         FYI: The relationship between bandwidth and Q is
         1/Q = 2*sinh(ln(2)/2*BW*w0/sin(w0))     (digital filter w BLT)
         or   1/Q = 2*sinh(ln(2)/2*BW)             (analog filter prototype)

         The relationship between shelf slope and Q is
         1/Q = sqrt((A + 1/A)*(1/S - 1) + 2)
         */

        var coeff;

        switch (this.type) {
            case DSP.LPF:       // H(s) = 1 / (s^2 + s/Q + 1)
                this.b0 =  (1 - cosw0)/2;
                this.b1 =   1 - cosw0;
                this.b2 =  (1 - cosw0)/2;
                this.a0 =   1 + alpha;
                this.a1 =  -2 * cosw0;
                this.a2 =   1 - alpha;
                break;

            case DSP.HPF:       // H(s) = s^2 / (s^2 + s/Q + 1)
                this.b0 =  (1 + cosw0)/2;
                this.b1 = -(1 + cosw0);
                this.b2 =  (1 + cosw0)/2;
                this.a0 =   1 + alpha;
                this.a1 =  -2 * cosw0;
                this.a2 =   1 - alpha;
                break;

            case DSP.BPF_CONSTANT_SKIRT:       // H(s) = s / (s^2 + s/Q + 1)  (constant skirt gain, peak gain = Q)
                this.b0 =   sinw0/2;
                this.b1 =   0;
                this.b2 =  -sinw0/2;
                this.a0 =   1 + alpha;
                this.a1 =  -2*cosw0;
                this.a2 =   1 - alpha;
                break;

            case DSP.BPF_CONSTANT_PEAK:       // H(s) = (s/Q) / (s^2 + s/Q + 1)      (constant 0 dB peak gain)
                this.b0 =   alpha;
                this.b1 =   0;
                this.b2 =  -alpha;
                this.a0 =   1 + alpha;
                this.a1 =  -2*cosw0;
                this.a2 =   1 - alpha;
                break;

            case DSP.NOTCH:     // H(s) = (s^2 + 1) / (s^2 + s/Q + 1)
                this.b0 =   1;
                this.b1 =  -2*cosw0;
                this.b2 =   1;
                this.a0 =   1 + alpha;
                this.a1 =  -2*cosw0;
                this.a2 =   1 - alpha;
                break;

            case DSP.APF:       // H(s) = (s^2 - s/Q + 1) / (s^2 + s/Q + 1)
                this.b0 =   1 - alpha;
                this.b1 =  -2*cosw0;
                this.b2 =   1 + alpha;
                this.a0 =   1 + alpha;
                this.a1 =  -2*cosw0;
                this.a2 =   1 - alpha;
                break;

            case DSP.PEAKING_EQ:  // H(s) = (s^2 + s*(A/Q) + 1) / (s^2 + s/(A*Q) + 1)
                this.b0 =   1 + alpha*A;
                this.b1 =  -2*cosw0;
                this.b2 =   1 - alpha*A;
                this.a0 =   1 + alpha/A;
                this.a1 =  -2*cosw0;
                this.a2 =   1 - alpha/A;
                break;

            case DSP.LOW_SHELF:   // H(s) = A * (s^2 + (sqrt(A)/Q)*s + A)/(A*s^2 + (sqrt(A)/Q)*s + 1)
                coeff = sinw0 * Math.sqrt( (A^2 + 1)*(1/this.S - 1) + 2*A );
                this.b0 =    A*((A+1) - (A-1)*cosw0 + coeff);
                this.b1 =  2*A*((A-1) - (A+1)*cosw0);
                this.b2 =    A*((A+1) - (A-1)*cosw0 - coeff);
                this.a0 =       (A+1) + (A-1)*cosw0 + coeff;
                this.a1 =   -2*((A-1) + (A+1)*cosw0);
                this.a2 =       (A+1) + (A-1)*cosw0 - coeff;
                break;

            case DSP.HIGH_SHELF:   // H(s) = A * (A*s^2 + (sqrt(A)/Q)*s + 1)/(s^2 + (sqrt(A)/Q)*s + A)
                coeff = sinw0 * Math.sqrt( (A^2 + 1)*(1/this.S - 1) + 2*A );
                this.b0 =    A*((A+1) + (A-1)*cosw0 + coeff);
                this.b1 = -2*A*((A-1) + (A+1)*cosw0);
                this.b2 =    A*((A+1) + (A-1)*cosw0 - coeff);
                this.a0 =       (A+1) - (A-1)*cosw0 + coeff;
                this.a1 =    2*((A-1) - (A+1)*cosw0);
                this.a2 =       (A+1) - (A-1)*cosw0 - coeff;
                break;
        }

        this.b0a0 = this.b0/this.a0;
        this.b1a0 = this.b1/this.a0;
        this.b2a0 = this.b2/this.a0;
        this.a1a0 = this.a1/this.a0;
        this.a2a0 = this.a2/this.a0;
    };

    this.process = function(buffer) {
        //y[n] = (b0/a0)*x[n] + (b1/a0)*x[n-1] + (b2/a0)*x[n-2]
        //       - (a1/a0)*y[n-1] - (a2/a0)*y[n-2]

        var len = buffer.length;
        var output = new Float32Array(len);

        for ( var i=0; i<buffer.length; i++ ) {
            output[i] = this.b0a0*buffer[i] + this.b1a0*this.x_1_l + this.b2a0*this.x_2_l - this.a1a0*this.y_1_l - this.a2a0*this.y_2_l;
            this.y_2_l = this.y_1_l;
            this.y_1_l = output[i];
            this.x_2_l = this.x_1_l;
            this.x_1_l = buffer[i];
        }

        return output;
    };

    this.processStereo = function(buffer) {
        //y[n] = (b0/a0)*x[n] + (b1/a0)*x[n-1] + (b2/a0)*x[n-2]
        //       - (a1/a0)*y[n-1] - (a2/a0)*y[n-2]

        var len = buffer.length;
        var output = new Float32Array(len);

        for (var i = 0; i < len/2; i++) {
            output[2*i] = this.b0a0*buffer[2*i] + this.b1a0*this.x_1_l + this.b2a0*this.x_2_l - this.a1a0*this.y_1_l - this.a2a0*this.y_2_l;
            this.y_2_l = this.y_1_l;
            this.y_1_l = output[2*i];
            this.x_2_l = this.x_1_l;
            this.x_1_l = buffer[2*i];

            output[2*i+1] = this.b0a0*buffer[2*i+1] + this.b1a0*this.x_1_r + this.b2a0*this.x_2_r - this.a1a0*this.y_1_r - this.a2a0*this.y_2_r;
            this.y_2_r = this.y_1_r;
            this.y_1_r = output[2*i+1];
            this.x_2_r = this.x_1_r;
            this.x_1_r = buffer[2*i+1];
        }

        return output;
    };
}

/*
 *  Magnitude to decibels
 *
 *  Created by Ricard Marxer <email@ricardmarxer.com> on 2010-05-23.
 *  Copyright 2010 Ricard Marxer. All rights reserved.
 *
 *  @buffer array of magnitudes to convert to decibels
 *
 *  @returns the array in decibels
 *
 */
DSP.mag2db = function(buffer) {
    var minDb = -120;
    var minMag = Math.pow(10.0, minDb / 20.0);

    var log = Math.log;
    var max = Math.max;

    var result = Float32Array(buffer.length);
    for (var i=0; i<buffer.length; i++) {
        result[i] = 20.0*log(max(buffer[i], minMag));
    }

    return result;
};

/*
 *  Frequency response
 *
 *  Created by Ricard Marxer <email@ricardmarxer.com> on 2010-05-23.
 *  Copyright 2010 Ricard Marxer. All rights reserved.
 *
 *  Calculates the frequency response at the given points.
 *
 *  @b b coefficients of the filter
 *  @a a coefficients of the filter
 *  @w w points (normally between -PI and PI) where to calculate the frequency response
 *
 *  @returns the frequency response in magnitude
 *
 */
DSP.freqz = function(b, a, w) {
    var i, j;

    if (!w) {
        w = Float32Array(200);
        for (i=0;i<w.length; i++) {
            w[i] = DSP.TWO_PI/w.length * i - Math.PI;
        }
    }

    var result = Float32Array(w.length);

    var sqrt = Math.sqrt;
    var cos = Math.cos;
    var sin = Math.sin;

    for (i=0; i<w.length; i++) {
        var numerator = {real:0.0, imag:0.0};
        for (j=0; j<b.length; j++) {
            numerator.real += b[j] * cos(-j*w[i]);
            numerator.imag += b[j] * sin(-j*w[i]);
        }

        var denominator = {real:0.0, imag:0.0};
        for (j=0; j<a.length; j++) {
            denominator.real += a[j] * cos(-j*w[i]);
            denominator.imag += a[j] * sin(-j*w[i]);
        }

        result[i] =  sqrt(numerator.real*numerator.real + numerator.imag*numerator.imag) / sqrt(denominator.real*denominator.real + denominator.imag*denominator.imag);
    }

    return result;
};

/*
 *  Graphical Equalizer
 *
 *  Implementation of a graphic equalizer with a configurable bands-per-octave
 *  and minimum and maximum frequencies
 *
 *  Created by Ricard Marxer <email@ricardmarxer.com> on 2010-05-23.
 *  Copyright 2010 Ricard Marxer. All rights reserved.
 *
 */
function GraphicalEq(sampleRate) {
    this.FS = sampleRate;
    this.minFreq = 40.0;
    this.maxFreq = 16000.0;

    this.bandsPerOctave = 1.0;

    this.filters = [];
    this.freqzs = [];

    this.calculateFreqzs = true;

    this.recalculateFilters = function() {
        var bandCount = Math.round(Math.log(this.maxFreq/this.minFreq) * this.bandsPerOctave/ Math.LN2);

        this.filters = [];
        for (var i=0; i<bandCount; i++) {
            var freq = this.minFreq*(Math.pow(2, i/this.bandsPerOctave));
            var newFilter = new Biquad(DSP.PEAKING_EQ, this.FS);
            newFilter.setDbGain(0);
            newFilter.setBW(1/this.bandsPerOctave);
            newFilter.setF0(freq);
            this.filters[i] = newFilter;
            this.recalculateFreqz(i);
        }
    };

    this.setMinimumFrequency = function(freq) {
        this.minFreq = freq;
        this.recalculateFilters();
    };

    this.setMaximumFrequency = function(freq) {
        this.maxFreq = freq;
        this.recalculateFilters();
    };

    this.setBandsPerOctave = function(bands) {
        this.bandsPerOctave = bands;
        this.recalculateFilters();
    };

    this.setBandGain = function(bandIndex, gain) {
        if (bandIndex < 0 || bandIndex > (this.filters.length-1)) {
            throw "The band index of the graphical equalizer is out of bounds.";
        }

        if (!gain) {
            throw "A gain must be passed.";
        }

        this.filters[bandIndex].setDbGain(gain);
        this.recalculateFreqz(bandIndex);
    };

    this.recalculateFreqz = function(bandIndex) {
        if (!this.calculateFreqzs) {
            return;
        }

        if (bandIndex < 0 || bandIndex > (this.filters.length-1)) {
            throw "The band index of the graphical equalizer is out of bounds. " + bandIndex + " is out of [" + 0 + ", " + this.filters.length-1 + "]";
        }

        if (!this.w) {
            this.w = Float32Array(400);
            for (var i=0; i<this.w.length; i++) {
                this.w[i] = Math.PI/this.w.length * i;
            }
        }

        var b = [this.filters[bandIndex].b0, this.filters[bandIndex].b1, this.filters[bandIndex].b2];
        var a = [this.filters[bandIndex].a0, this.filters[bandIndex].a1, this.filters[bandIndex].a2];

        this.freqzs[bandIndex] = DSP.mag2db(DSP.freqz(b, a, this.w));
    };

    this.process = function(buffer) {
        var output = buffer;

        for (var i = 0; i < this.filters.length; i++) {
            output = this.filters[i].process(output);
        }

        return output;
    };

    this.processStereo = function(buffer) {
        var output = buffer;

        for (var i = 0; i < this.filters.length; i++) {
            output = this.filters[i].processStereo(output);
        }

        return output;
    };
}

/**
 * MultiDelay effect by Almer Thie (http://code.almeros.com).
 * Copyright 2010 Almer Thie. All rights reserved.
 * Example: http://code.almeros.com/code-examples/delay-firefox-audio-api/
 *
 * This is a delay that feeds it's own delayed signal back into its circular
 * buffer. Also known as a CombFilter.
 *
 * Compatible with interleaved stereo (or more channel) buffers and
 * non-interleaved mono buffers.
 *
 * @param {Number} maxDelayInSamplesSize Maximum possible delay in samples (size of circular buffer)
 * @param {Number} delayInSamples Initial delay in samples
 * @param {Number} masterVolume Initial master volume. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
 * @param {Number} delayVolume Initial feedback delay volume. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
 *
 * @constructor
 */
function MultiDelay(maxDelayInSamplesSize, delayInSamples, masterVolume, delayVolume) {
    this.delayBufferSamples   = new Float32Array(maxDelayInSamplesSize); // The maximum size of delay
    this.delayInputPointer     = delayInSamples;
    this.delayOutputPointer   = 0;

    this.delayInSamples   = delayInSamples;
    this.masterVolume     = masterVolume;
    this.delayVolume     = delayVolume;
}

/**
 * Change the delay time in samples.
 *
 * @param {Number} delayInSamples Delay in samples
 */
MultiDelay.prototype.setDelayInSamples = function (delayInSamples) {
    this.delayInSamples = delayInSamples;

    this.delayInputPointer = this.delayOutputPointer + delayInSamples;

    if (this.delayInputPointer >= this.delayBufferSamples.length-1) {
        this.delayInputPointer = this.delayInputPointer - this.delayBufferSamples.length;
    }
};

/**
 * Change the master volume.
 *
 * @param {Number} masterVolume Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
 */
MultiDelay.prototype.setMasterVolume = function(masterVolume) {
    this.masterVolume = masterVolume;
};

/**
 * Change the delay feedback volume.
 *
 * @param {Number} delayVolume Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
 */
MultiDelay.prototype.setDelayVolume = function(delayVolume) {
    this.delayVolume = delayVolume;
};

/**
 * Process a given interleaved or mono non-interleaved float value Array and adds the delayed audio.
 *
 * @param {Array} samples Array containing Float values or a Float32Array
 *
 * @returns A new Float32Array interleaved or mono non-interleaved as was fed to this function.
 */
MultiDelay.prototype.process = function(samples) {
    // NB. Make a copy to put in the output samples to return.
    var outputSamples = new Float32Array(samples.length);

    for (var i=0; i<samples.length; i++) {
        // delayBufferSamples could contain initial NULL's, return silence in that case
        var delaySample = (this.delayBufferSamples[this.delayOutputPointer] === null ? 0.0 : this.delayBufferSamples[this.delayOutputPointer]);

        // Mix normal audio data with delayed audio
        var sample = (delaySample * this.delayVolume) + samples[i];

        // Add audio data with the delay in the delay buffer
        this.delayBufferSamples[this.delayInputPointer] = sample;

        // Return the audio with delay mix
        outputSamples[i] = sample * this.masterVolume;

        // Manage circulair delay buffer pointers
        this.delayInputPointer++;
        if (this.delayInputPointer >= this.delayBufferSamples.length-1) {
            this.delayInputPointer = 0;
        }

        this.delayOutputPointer++;
        if (this.delayOutputPointer >= this.delayBufferSamples.length-1) {
            this.delayOutputPointer = 0;
        }
    }

    return outputSamples;
};

/**
 * SingleDelay effect by Almer Thie (http://code.almeros.com).
 * Copyright 2010 Almer Thie. All rights reserved.
 * Example: See usage in Reverb class
 *
 * This is a delay that does NOT feeds it's own delayed signal back into its
 * circular buffer, neither does it return the original signal. Also known as
 * an AllPassFilter(?).
 *
 * Compatible with interleaved stereo (or more channel) buffers and
 * non-interleaved mono buffers.
 *
 * @param {Number} maxDelayInSamplesSize Maximum possible delay in samples (size of circular buffer)
 * @param {Number} delayInSamples Initial delay in samples
 * @param {Number} delayVolume Initial feedback delay volume. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
 *
 * @constructor
 */

function SingleDelay(maxDelayInSamplesSize, delayInSamples, delayVolume) {
    this.delayBufferSamples = new Float32Array(maxDelayInSamplesSize); // The maximum size of delay
    this.delayInputPointer  = delayInSamples;
    this.delayOutputPointer = 0;

    this.delayInSamples     = delayInSamples;
    this.delayVolume        = delayVolume;
}

/**
 * Change the delay time in samples.
 *
 * @param {Number} delayInSamples Delay in samples
 */
SingleDelay.prototype.setDelayInSamples = function(delayInSamples) {
    this.delayInSamples = delayInSamples;
    this.delayInputPointer = this.delayOutputPointer + delayInSamples;

    if (this.delayInputPointer >= this.delayBufferSamples.length-1) {
        this.delayInputPointer = this.delayInputPointer - this.delayBufferSamples.length;
    }
};

/**
 * Change the return signal volume.
 *
 * @param {Number} delayVolume Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
 */
SingleDelay.prototype.setDelayVolume = function(delayVolume) {
    this.delayVolume = delayVolume;
};

/**
 * Process a given interleaved or mono non-interleaved float value Array and
 * returns the delayed audio.
 *
 * @param {Array} samples Array containing Float values or a Float32Array
 *
 * @returns A new Float32Array interleaved or mono non-interleaved as was fed to this function.
 */
SingleDelay.prototype.process = function(samples) {
    // NB. Make a copy to put in the output samples to return.
    var outputSamples = new Float32Array(samples.length);

    for (var i=0; i<samples.length; i++) {

        // Add audio data with the delay in the delay buffer
        this.delayBufferSamples[this.delayInputPointer] = samples[i];

        // delayBufferSamples could contain initial NULL's, return silence in that case
        var delaySample = this.delayBufferSamples[this.delayOutputPointer];

        // Return the audio with delay mix
        outputSamples[i] = delaySample * this.delayVolume;

        // Manage circulair delay buffer pointers
        this.delayInputPointer++;

        if (this.delayInputPointer >= this.delayBufferSamples.length-1) {
            this.delayInputPointer = 0;
        }

        this.delayOutputPointer++;

        if (this.delayOutputPointer >= this.delayBufferSamples.length-1) {
            this.delayOutputPointer = 0;
        }
    }

    return outputSamples;
};

/**
 * Reverb effect by Almer Thie (http://code.almeros.com).
 * Copyright 2010 Almer Thie. All rights reserved.
 * Example: http://code.almeros.com/code-examples/reverb-firefox-audio-api/
 *
 * This reverb consists of 6 SingleDelays, 6 MultiDelays and an IIRFilter2
 * for each of the two stereo channels.
 *
 * Compatible with interleaved stereo buffers only!
 *
 * @param {Number} maxDelayInSamplesSize Maximum possible delay in samples (size of circular buffers)
 * @param {Number} delayInSamples Initial delay in samples for internal (Single/Multi)delays
 * @param {Number} masterVolume Initial master volume. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
 * @param {Number} mixVolume Initial reverb signal mix volume. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
 * @param {Number} delayVolume Initial feedback delay volume for internal (Single/Multi)delays. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
 * @param {Number} dampFrequency Initial low pass filter frequency. 0 to 44100 (depending on your maximum sampling frequency)
 *
 * @constructor
 */
function Reverb(maxDelayInSamplesSize, delayInSamples, masterVolume, mixVolume, delayVolume, dampFrequency) {
    this.delayInSamples   = delayInSamples;
    this.masterVolume     = masterVolume;
    this.mixVolume       = mixVolume;
    this.delayVolume     = delayVolume;
    this.dampFrequency     = dampFrequency;

    this.NR_OF_MULTIDELAYS = 6;
    this.NR_OF_SINGLEDELAYS = 6;

    this.LOWPASSL = new IIRFilter2(DSP.LOWPASS, dampFrequency, 0, 44100);
    this.LOWPASSR = new IIRFilter2(DSP.LOWPASS, dampFrequency, 0, 44100);

    this.singleDelays = [];

    var i, delayMultiply;

    for (i = 0; i < this.NR_OF_SINGLEDELAYS; i++) {
        delayMultiply = 1.0 + (i/7.0); // 1.0, 1.1, 1.2...
        this.singleDelays[i] = new SingleDelay(maxDelayInSamplesSize, Math.round(this.delayInSamples * delayMultiply), this.delayVolume);
    }

    this.multiDelays = [];

    for (i = 0; i < this.NR_OF_MULTIDELAYS; i++) {
        delayMultiply = 1.0 + (i/10.0); // 1.0, 1.1, 1.2...
        this.multiDelays[i] = new MultiDelay(maxDelayInSamplesSize, Math.round(this.delayInSamples * delayMultiply), this.masterVolume, this.delayVolume);
    }
}

/**
 * Change the delay time in samples as a base for all delays.
 *
 * @param {Number} delayInSamples Delay in samples
 */
Reverb.prototype.setDelayInSamples = function (delayInSamples){
    this.delayInSamples = delayInSamples;

    var i, delayMultiply;

    for (i = 0; i < this.NR_OF_SINGLEDELAYS; i++) {
        delayMultiply = 1.0 + (i/7.0); // 1.0, 1.1, 1.2...
        this.singleDelays[i].setDelayInSamples( Math.round(this.delayInSamples * delayMultiply) );
    }

    for (i = 0; i < this.NR_OF_MULTIDELAYS; i++) {
        delayMultiply = 1.0 + (i/10.0); // 1.0, 1.1, 1.2...
        this.multiDelays[i].setDelayInSamples( Math.round(this.delayInSamples * delayMultiply) );
    }
};

/**
 * Change the master volume.
 *
 * @param {Number} masterVolume Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
 */
Reverb.prototype.setMasterVolume = function (masterVolume){
    this.masterVolume = masterVolume;
};

/**
 * Change the reverb signal mix level.
 *
 * @param {Number} mixVolume Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
 */
Reverb.prototype.setMixVolume = function (mixVolume){
    this.mixVolume = mixVolume;
};

/**
 * Change all delays feedback volume.
 *
 * @param {Number} delayVolume Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
 */
Reverb.prototype.setDelayVolume = function (delayVolume){
    this.delayVolume = delayVolume;

    var i;

    for (i = 0; i<this.NR_OF_SINGLEDELAYS; i++) {
        this.singleDelays[i].setDelayVolume(this.delayVolume);
    }

    for (i = 0; i<this.NR_OF_MULTIDELAYS; i++) {
        this.multiDelays[i].setDelayVolume(this.delayVolume);
    }
};

/**
 * Change the Low Pass filter frequency.
 *
 * @param {Number} dampFrequency low pass filter frequency. 0 to 44100 (depending on your maximum sampling frequency)
 */
Reverb.prototype.setDampFrequency = function (dampFrequency){
    this.dampFrequency = dampFrequency;

    this.LOWPASSL.set(dampFrequency, 0);
    this.LOWPASSR.set(dampFrequency, 0);
};

/**
 * Process a given interleaved float value Array and copies and adds the reverb signal.
 *
 * @param {Array} samples Array containing Float values or a Float32Array
 *
 * @returns A new Float32Array interleaved buffer.
 */
Reverb.prototype.process = function (interleavedSamples){
    // NB. Make a copy to put in the output samples to return.
    var outputSamples = new Float32Array(interleavedSamples.length);

    // Perform low pass on the input samples to mimick damp
    var leftRightMix = DSP.deinterleave(interleavedSamples);
    this.LOWPASSL.process( leftRightMix[DSP.LEFT] );
    this.LOWPASSR.process( leftRightMix[DSP.RIGHT] );
    var filteredSamples = DSP.interleave(leftRightMix[DSP.LEFT], leftRightMix[DSP.RIGHT]);

    var i;

    // Process MultiDelays in parallel
    for (i = 0; i<this.NR_OF_MULTIDELAYS; i++) {
        // Invert the signal of every even multiDelay
        outputSamples = DSP.mixSampleBuffers(outputSamples, this.multiDelays[i].process(filteredSamples), 2%i === 0, this.NR_OF_MULTIDELAYS);
    }

    // Process SingleDelays in series
    var singleDelaySamples = new Float32Array(outputSamples.length);
    for (i = 0; i<this.NR_OF_SINGLEDELAYS; i++) {
        // Invert the signal of every even singleDelay
        singleDelaySamples = DSP.mixSampleBuffers(singleDelaySamples, this.singleDelays[i].process(outputSamples), 2%i === 0, 1);
    }

    // Apply the volume of the reverb signal
    for (i = 0; i<singleDelaySamples.length; i++) {
        singleDelaySamples[i] *= this.mixVolume;
    }

    // Mix the original signal with the reverb signal
    outputSamples = DSP.mixSampleBuffers(singleDelaySamples, interleavedSamples, 0, 1);

    // Apply the master volume to the complete signal
    for (i = 0; i<outputSamples.length; i++) {
        outputSamples[i] *= this.masterVolume;
    }

    return outputSamples;
};


function benchmark(func, loopCount) {
    loopCount = loopCount || 10000;

    var start = Date.now();

    for (var i = 0; i < loopCount; i++) {
        func();
    }

    var end = Date.now();
    return end - start;
}


var FREQUENCY;
var bufferSize = 2048;
var sampleRate = 44100;
var frequency = FREQUENCY || 440;

var fft = new FFT(bufferSize, sampleRate);
var osc = new Oscillator(DSP.SAW, frequency, 1.0, bufferSize, sampleRate);
var signal = osc.generate();

var duration = benchmark(function() { fft.forward(signal); },2);

var peakBand = 0;

for (var i = 0; i < fft.spectrum.length; i++) {
    peakBand = (fft.spectrum[i] > fft.spectrum[peakBand]) ? i : peakBand;
}

var peakFreq = fft.getBandFrequency(fft.peakBand);

console.log("Detected peak: " + peakFreq + " Hz (error " + Math.abs(peakFreq - frequency) + " Hz)");
console.log("10000 FFTs: " + (duration) + " ms (" + ((duration) / 10000) + "ms per FFT)\n");